AM Stereo Tuner/FM Stereo IF Receiver Written by BuSan Wednesday, 08 August 2007 This application combines a Sanyo LA1832M withthe Motorola MC13028A AM Stereo decoder IC. The LA FM IF, FM multiplex detection, AM tuning, and the AM IF functions. The MC13028A provides the AM Ster well as Left and Right audio outputs. An MC145151 synthesizer provides the frequency control of the local o contained within the LA1832. Frequency selection is by means of a switch array attached to the synthesizer. The MC13028A is designed as a low voltage, low cost decoder for the C–QUAM AM Stereo technology and compatible with existing monaural AM transmissions. The IC requires relatively few, inexpensive external pa full featured C–QUAM AM Stereo implementation. The layout is straightforward and should produce excelle performance. This device performs the function of IF amplification, AGC, modulation detection, pilot tone de quality inspection, and left and right audio output matrix operation. The IC is targeted for use in portable and radio applications. The amplified C–QUAM IF signal is fed simultaneously to the envelope detector circuit, and to a C–QUAM The envelope detector provides the L+R (mono) signal output which is fed to the stereo matrix. In the conver QUAM signal is restored to a Quam signal. This is accomplished by dividing the C–QUAM IF signal by the d Φ term. The cosΦ term is derived from the phase modulated IF signal in an active feedback loop. Cosine Φ is comparing the envelope detector and the in–phase detector outputs in the high speed comparator/feedback loo extracted from the I detector output and is actively transferred through feedback to the output of the comparat the comparator is in turn fed to the control input of the divider, thus closing the feedback loop of the converte process, the cos Φ term is removed from the divider IF output, thus allowing direct detection of the L–R by th detector. The audio outputs from both the envelope and the L–R detectors are first filtered to minimize the se the IF signal. Then they are fed into a matrix circuit where the Left channel and the Right channel outputs can Pins 15 and 16. (The outputs from the I and Q detectors are also filtered similarly.) At this time, a stereo indic circuit, which can sink up to 10 mA, is also enabled. The stereo output will occur if the input IF signal is: larg threshold level, not too noisy, and if a proper pilot tone is present. If these three conditions are not met, the bl begin to force monaural operation at that time. A blend circuit is included in this design because conditions occur during field use that can cause input signal fluctuation, strong unwanted co–channel or power line interference, and/or multi–path or re–radiation. When conditions occur, rapid switching between stereo and mono might occur, or the stereo quality might be degrad sound displeasing. Since these conditions could be annoying to the normal listener, the stereo information is b monaural output. This circuit action creates a condition for listening where these aberrant effects are better to consumer. Intentional mono operation is a feature sometimes required in receiver designs. There are several ways in whi this feat. First, a resistor from Pin 10 to ground can be switched into the circuit. A value of 1.0 k is adequate a schematic in Figure 18. A second method to force the decoder into mono is simply to shunt Pin 10 to ground transistor (collector to Pin 10, emitter to ground), where the base lead is held electrically “high” to initiate the A third method to force a mono condition upon the decoder is to shunt Pin 8 of the decoder to ground through transistor as described above. Effectively, this operation discharges the blend capacitor (10 μF), and the blend over internally forcing the decoder into mono. This third method does not necessarily require extra specific p mono function as the first two examples do. The reason for this is that most electronically tuned receiver de audio muting function during turn on/turn off, tuning/scanning, or band switching (FM to AM). When the mu designed into an AM Stereo receiver, it also should include a blend capacitor reset (discharge) function which is this case by the use of an NPN transistor shunting Pin 8 to ground, (thus making the addition of a forced mon “free”). The purpose of the blend reset during muting is to re–initialize the decoder back into the “fast lock” m stereo operation can be attained much quicker after any of the interruptive activities mentioned earlier, (i.e. tu etc.). The VCO in this IC is a phase shift oscillator type design that operates with a ceramic resonator at eight times or 3.60 MHz. With IF input levels below the stereo threshold level, the oscillator is not operational. This featu eliminate audio tweets under low level, noisy input conditions. The following information provides circuit function, part number, and the manufacturer’s name for special pa their schematic symbol. Where the part is not limited to a single source, a description sufficient to select a par U1 IC – AM Stereo Decoder MC13028AD by Motorola U2 IC – AM/FM IF and Multiplex Tuner LA1832M by Sanyo U3 IC – Frequency Synthesizer MC145151DW2 by Motorola T1 AM IF Coil A7NRES–11148N by TOKO F1 AM IF Ceramic Filter SFG450F by Murata F2 FM IF Detector Resonator CDA10.7MG46A by Murata F3 FM Multiplex Decoder Resonator CSB456F15 by Murata F4 AM Tuner Block BL–70 by Korin Giken X1 10.24 MHz Crystal, Fundamental Mode, AT Cut, 18 pF Load Cap, 35 Ω maximum series R. HC–18/U Holder X2 3.6 MHz AM Stereo Decoder Resonator CSA3.60MGF108 by Murata S5 8 SPST DIP Switch Figure 1. AM STEREO TUNER/FM STEREO IF Circuit with MC13028, MC145151 and L The secondary is tuned by a varactor which is controlled by a dc voltag synthesizer circuit.O. and also to the input. is determined. The IF signal returns to U2 through Pin 4. A buffered output from the L. It is through this reactance of the L. The reactance of this oscillator tank is coupled back to Pin 23.Figure 2. That coil applies the signal to Pin 21 coil is connected from Pin 23 to VCC. emerges at Pin 24. The mixer output at Pin 2 is applied to the IF coil T1. Pin 4 of the AM Stereo decoder (U1 . AM STEREO TUNER/FM STEREO IF PCB with MC13028. Coil T1 provides the correct impedance to drive the cer filter F1. This signal is routed to Pin 1 of (U3). thus completing the frequency control loop. An AM Stereo applied to Pin 2 of the RF coil contained within the BL–70 tuning block. MC145151 and LA The LA1832 tuner IC (U2) is set for AM operation by switch S2 connecting Pin 12 to ground.O. the IF frequency oscillator value within the AM band can be represented by a binary number.0 kΩ. a computation must be made in order to ascertain the divide ratio to input via the switch array. Each binary bit represents a switch setting where a “1” is an open switch and a “0” is a c most significant bit represents switch 8 which is connected to Pin 18. Divid yields the number 152. the response of the speakers or earphones should also be considered.filter F1 is designed to operate into a load resistance of 2. and they must all be considered when trying to approximate the NRSC de–emphasis re output transformer (IF coil. Thus the switches are set to: Switch Configuration The MC13025 is the complementary ETR® Electronically Tuned Radio front–end for the second generation MC13022 C–QUAM® AM stereo IF and decoder. The other switches connec between and in order. Pins 15 and 16 are also good locations for the insertion of simple RC filters that are used to comply with the U NRSC requirement for the shape of the overall receiver audio response. There are many design factors that af the receiver response. To tune to a specific RF frequency. To illustrate.) The local oscillator frequency is 1070 kHz plus 450 kHz which equals 1520 kHz. Number 1 connects to Pin 11 of U3 and number 8 connects to Pin 18. (This frequency was used to test the cir described further on. The frequency to which the test circuit will tune is set by the eight binary switches contained in the S5 assemb from 1 to 8. When de audio response shape. The MC13025 provides a high dynamic range mixer. The binary number for 152 is 10011000. The stereo outputs exit from Pins 15 and 16 of U1. The design amplitudes of the audio outputs will vary acco values used for the resistors to ground at Pins 15 and 16 of the decoder. the numbers chosen in this data sheet are reflective of those required to set the general industr of audio outputs in receiver designs. The divide ratio is simply the eight digit binary equivalent number for the local oscillato divided by 10 kHz. voltage controlled oscillator. T1). consider the setting for an input frequency of 1070 kHz. This load is provided at Pin 4 of U2. Each individual switch is a SPST type. The local oscillator frequency is the desired RF frequency plus 450 kHz. and first IF that with . and ceramic filter probably have the greatest contribution to the frequency resp filter can be tailored from its rated response by the choice of transformer impedance and bandwidth. While the values chosen for RO are lef of the designer. This system in turn may drive a dual channel audio processor and high power amplifiers for car radio or home stereo applications. MC13025 and MC13022 Cascode RF ETR AM Stereo Receiver Application Circuit Blend and Noise Reduction Although AM stereo does not have the extreme difference in S/N between mono and stereo that FM does (typically less than 3. Some forms of interference such as co–channel have a large L–R component that makes them more annoying than would ordinarily be expected for the measured level. blending smoothly to monaural.the MC13022 and synthesizer form a complete digitally controlled AM stereo tuner system.0 dB versus greater than 20 dB for FM). The stop sense/AGC function has been internally connected to the output notch filter control. Figure 1. The . table radios and component stereo systems. Other applications include portable radio “boom boxes”. The MC13022A has 10 dB more audio output and a CMOS compatible logic level output (Pin 15) for stop sense. Functionally. the MC13022A and MC13022 are very similar. sudden switching between mono and stereo is quite apparent. The MC13022A measures the interference level and reduces L–R as interference increases. Response is controlled by Pin 6 MC13022A for automatic audio bandwidth control as a function of signal strength. noisy signals. By using a twin–T filter with variable feedback to the normally grounded center leg.. Pin 6 drives a comparator which has a 1. Pin 15. notch filters are necessary with any wide–band AM radio to eliminate the 10 kHz whistle from adjacent stations.7 V reference. Pins 6 and 15 will go low. This widens the IF bandwidth by decreasing the loaded Q of the input coupling coil as signal strength increases.pilot indicator remains on as long as a pilot signal is detected. a variable Q notch filter is formed that provides both the 10 kHz notch and variable high frequency rolloff functions. high frequency rolloff greatly improves the sound. Signal Strength A dc voltage proportional to the log of signal strength is provided at Pin 6 MC13022A. Therefore the comparator output. below 0. The interference detection prevents stopping on many unlistenable stations.3 V. a feature particularly useful at night when many frequencies may have strong signals from multiple co–channel stations. is low if Pin 6 is <1. Normal operation is above 2.7 V and high if Pin 6 is >1.2 V as shown is Figure NO TAG. This can be used for signal strength indication. Supply Voltage Ranger= +10V.. and it directly controls the post detection filter. This would typically be done from the mute line in a frequency synthesizer.+18V Supply Current whit LED= 30 mA Stereo Separation= 45 dB . IF Bandwidth Control IF AGC attenuates the signal by shunting the signal at the IF input. Also. even when interference is severe.7 V. Stop Sense The signal strength information is multiplexed with the stop sense signal. If at any time Pin 23 is low and there is either no signal in the IF or a noisy signal of a predetermined interference level. The stop sense is activated when scanning by externally pulling the blend.. to minimize annoying pilot light flickering.. Conventional tone controls do not attenuate the highs sufficiently to control noise without also significantly affecting the mid–range. Typical range of response is shown in Figure NO TAG. Pin 23 MC13022A. This low can be used to tell the frequency synthesizer to immediately scan to the next channel. Post Detection Filtering With weak. 5Vrms-1KHZ = 0.3% Channel Separation= 40dB MC1310P: See Database Specification .H.T.4% Input Resistance= 45 K Output Resistance= 1.D at 0.3K SCA Rejection= 70 dB LM1800N: See Database Power supply =8V-14V Current with out LED =13 mA Input signal = 1V RMS. Output signal= 485 mV Distortion = 0. The FM waveform can be clipped at a low level without the loss of information. so the amplitude is constant. 17 November 2006 An FM waveform carries its information in the form of frequency. or limit the amplitude of the received waveform prior to frequency detection. Thus the information is held in the zero crossings. Additive noise has less of an effect on zero crossings than the amplitude. A simplified FM receiver is shown in figure 1a. This produces a constant waveform as an input to the discriminator. This clipping has the effect of introducing higher harmonic terms which are rejected by a pos-detection lowpass filter. .FM Receiver Written by BuSan Friday. Receivers therefore often clip. a more sophisticated system is shown in figure 1b. they work by converting the FM signal to AM then demodulate using conventional peak detectors. The differentiator effectively converts the FM signal into an AM signal. Foster-Seeley disciminator. Descriminators A block diagram of a descriminator is shown in figure 2. . Several circuits are used for demodulating FM signals slope detector. ratio detector. The first three are tuned circuit frequency discriminators. PLL demodulator and quadrature detector. The signal received is lfm(t) and is known to the receiver in the form.DEMODULATORS FM demodulators are frequency-dependant circuits that produce an output voltage that is directly proportional to the instantaneous frequency at its input. The differentiated FM signal is. the Foster-Seeley discriminator or also known as a phase shift demodulator. The main problem with introducing stereophonic transmission is the compatibility . and music that originated on the right side is reproduced only on the right speaker. The envelope is given by from which the signal s(t) can be found. see figure 4. 17 November 2006 Until 1961. see figure 3. all commercial FM broadcast-band transmissions were monophonic. 08 December 2006 ) FM Stereo Written by BuSan Friday. the information signal is spatially divided into two 50-Hz to 15-kHz audio channels (a left and a right). A requirement for a descriminator is that the transfer function be linear throughout the range of frequencies of the FM wave. When a differentiator is used like this it is called a slope detector or discriminator. The ratio detector has the property that it is immune to amplitude variations in its input signal. This is the simplest type of decriminator. Two descriminators can be used by subtracting the characteristic of one from a shifted version of itself. This method is called a balanced slope detector. Music that originated on the left side is reproduced only on the left speaker. Another way is to approximate the derivative by using the difference between two adjacent sample values of the waveform. The FCC authorized stereophonic transmission for the commercial FM broadcast band. so a preceeding limiter is not required. The Foster-Seeley circuit is easier to tune but must be preceeded by a separate limiter circuit to clip the amplitude before demodulating. Last Updated ( Friday. It has several disadvantages like poor linearity and difficulty in tuning.The envelope detector removes the sine term. this is possible because the slight changes in frequency are not detected by the envelope detector. With stereophonic transmission. The spectrum compromises the 50 Hz to 15 kHzstereo channel plus an additional stereo channel frequency division multiplexed into a composite baseband signal with a 19 kHz pilot. Mono receivers can demodulate the total baseband spectrum but only the 50 . The three channels are (1) the left (L) plus the right (R) audio channels.53 kHz L . separate the left and right audio channels and then feed them to their respective speakers. The process of multiplexing two audio signals is shown in figure 2. used only for FM stereo transmission. Stereophonic receivers must provide additional demodulation of the 23 . The spectrum shown in figure 1 is the standard spectrum used today. The L . (2) the left plus the inverted right audio channels. SCA transmission occupy the 60 .with monophonic receivers. .15 kHz L + R channel is amplified and fed to all speakers.R audio channel amplitude modulates a 38 kHz subcarrier and produces the L .15 kHz passband.R stereo channel. which is a double-sideband suppressed carrier that occupies the 23 .R stereo channel.74 kHz frequency spectrum. and (3) the SCA subcarrier and its associated sidebands.53 kHz passband. The L + R stereo channel occupies the 0 . R stereo channel for demodulation purposes. Also for demodulation purposes. the baseband signal is fed into the stereo demodulator where the L and R audio channels are separated and then fed to their respective speakers.R audio information. The L + R and L . The L . the L + R stereo channel. The output of the discriminator is the total baseband spectrum that was shown in figure 1. Stereo Reception FM stereo receivers are identical to standard FM receivers up to the output of the audio detector stage. The matrix network combines the L + R and . Because there is a time delay introduced in the L . In the stereo section of the signal processor. The composite baseband signal is fed to the FM transmitter. which contains all of the original information from both the L and R audio channels. amplified and the fed to both the L and R speakers.R stereo channels and the 19 kHz pilot are separated from the composite baseband signal with filters.R signal path as it propagates through the balanced modulator. The L and R audio channels are combined in a matrix network to produce the L + R and the L . amplified and the fed to the L .R double-sideband signal is separated with a broadly tuned bandpass filter and then mixed with the recovered 38 kHz carrier in a balanced modulator to produce the L . where it modulates the main carrier. is simply filtered. multiplied by 2.Stereo Transmission Figure 3 shows a simplified block diagram for a stereo FM transmitter.R audio channel modulates a 38 kHz subcarrier and produces a 23 to 53 kHz L . The L + R stereo channel is filtered off by a low-pass filter with an upper cutoff frequency of 15 kHz.R audio channels.R stereo channel.R demodulator. In the mono section of the signal processor. Figure 4 shows a simplified block diagram for an FM receiver that has both mono and stereo audio outputs. a 19 kHz pilot is transmitted rather than the 38 kHz subcarrier because it is considerably more difficult to recover the 38 kHz subcarrier in the receiver. The L . The 19 kHz pilot is filtered with a high-Q bandpass filter. the L + R stereo channel must be artificially delayed somewhat to maintain phase integrity with the L . R audio channel is inverted and then added to the L + R audio channel.L . The ouput from the adder is The L . The output from the adder is . The L .R signals in such a way as to separate the L and R audio information signals. The block diagram for the stereo matrix decoder if shown in figure 5. which are fed to their respective deemphasis networks and speakers.R audio channel is added directly to the L + R audio channel. Frequency Modulation Principles Written by BuSan Friday. We define the instantaneous frequency of the signal to be Note that the power of the FM signal is a constant (with value A2). 17 November 2006 Frequency modulation of a carrier signal (with frequency fC) involves changing the frequency of this carrier based on some input signal s(t). The modulated carrier then contains the information in s(t). Figures 1 & 2 show some examples of frequency modulated signals and the corresponding inputs. where (t) is the varying phase angle of the signal. . This process is useful in radio communications because it is not practical to directly transmit audio frequencies through the air. The modulated carrier is of the form Acos [ (t)] . This analysis yields a Bessel function of . we can perform a Fourier series expansion of the FM signal to find its spectrum. ie this is a measure of the amount of modulation. Since the FM signal is periodic (for a sine wave input). then the phase angle is and the FM signal is We define as the modulation index of the FM signal.To examine the FM signal's spectrum. we will look at the various harmonics comprising the input signal. This gives us a measure of how much the frequency of the FM signal changes. If the input signal is s(t) = X cos 2 fmt. the magnitude of the ( +1)th harmonic is negligible. Demokratisasi Pemancar Radio Written by BuSan Friday. karena kita menggunakan sumber daya alam yang terbatas. Pada hari ini. oleh komunitas. sehingga ?tidak mungkin? menerima sebuah radio siaran baru kecuali dilakukan realokasi frekuensi. yaitu frekuensi radio. cerita menjadi lain. Pada media radio. untuk komunitas itu sendiri.the first kind. we use (t) = 2 fCt + kg(t). In fact. AM atau MW pada frekuensi 535-1705KHz . so we assume that there are in fact only b harmonics. This tells us that the magnitude of the harmonics is eventually decreasing (ie the function does not monotonicly decrease. yang praktis tidak menggunakan sumber daya alam yang terbatas. Media memungkinkan seseorang / sekelompok orang untuk mengekspresikan pendapatnya kepada banyak kalangan. Proses ini mungkin mudah di implementasikan di media cetak. For general signals with maximum frequency deviation f (ie f = max{kX}). Band radio siaran sebetulnya masih banyak. but with Thus. increasing the frequency of the input signal requires more bandwidth for transmission. we can see that phase modulation and frequency modulation are very similar. but it eventually gets smaller). but this is not necessarily the maximum frequency deviation. In fact. FM dengan alokasi frekuensi 88-108MHz dengan spasi 350KHz antar pemancar merupakan band favorit diantara seluruh band radio siaran yang ada. In this case. since this depends of the magnitude of the Fourier transform. collectively the are described as Angle Modulation . so the analysis presented applies to both. we can say that the bandwidth of the FM signal is approximately 2( fm + fm). we can say that BW 2( f + fm). 17 November 2006 Media adalah kunci strategis proses demokratisasi. Also. Kapling band FM di Jakarta bahkan naga-naganya sudah penuh. Media juga memudahkan interaksi dalam sebuah komunitas. Having done this analysis (skipping the messy mathematical details). This equation is known as Carson's Rule and can be written as BW 2(kX + fm). Phase Modulation involves changing the phase of the carrier signal based on the input g(t). This is the same as frequency modulation. Note that fm is the maximum frequency of the input signal (ie the input bandwidth is 2fm). We can thus see that the bandwidth of the FM signal depends on the size of the frequency deviations from the carrier frequency. The concept of phase modulation will be useful in later sections . which is also intuitive. 9MHz. Konsekuensinya.6-26. FM praktis penuh di beberapa kota besar. Pada gelombang pendek (SW). Sebetulnya pada hari ini teman-teman radio komunitas sudah mulai memancarkan diri di band FM dengan pemancar buatan sendirinya di Wanayasa.1MHz.05MHz.6-13. 11. bahkan versi yang berdaya rendah dapat dibuat beberapa ratus ribu rupiah.6MHz.9MHz. 7. 13. Gunakan keyword "pirate FM".85MHz.1-7. 5. 21. AM cukup baik untuk penggunaan radio siaran terutama di tepi laut. maka kepentingan banyak orang menjadi terkangkangi oleh satu . search engine Pertanyaan mendasar yang perlu dipikirkan bersama. peralatan pemancar AM & SW sebetulnya jauh lebih sederhana dan jauh lebih murah daripada pemancar FM.5517.495MHz.biasanya kosong karena tidak ada yang berminat memancar pada band ini.0MHz. Bagi pembaca yang penasaran dengan teknik pemancar FM.20MHz. muncul kebutuhan untuk membangun radio siaran komunitas. 17. warga masyarakat. Konsep radio siaran komunitas. 9.65-12. Pada hari ini radio siaran swasta niaga lebih banyak di fokuskan untuk usaha. penataan frekuensi agar kebutuhan komunitas ini menjadi terpenuhi? Jawaban sederhananya sebetulnya tidak sukar.2-3. bisnis & mencari fulus yang halal. 4.9-4. kebutuhan komunitas yang terbatas seperti komunitas sekolah. Yang perlu di garis bawahi. Berdasarkan kebutuhan tersebut.8MHz. sialnya. seperti www. yang akan menempuh jarak sangat jauh. benarkah keberadaan pemancar komunitas ini merupakan pemborosan frekuensi? Bagaimana disain alokasi frekuensi. Mereka umumnya tergabung pada Jaringan Radio Komunitas. komunitas nelayan. sangat terjangkau untuk sebuah sekolah atau kelompok warga untuk mulai memancar. Tentunya masih di tambah lagi siaran radio melalui satelit maupun Internet.06MHz. tidak lebih dari satu (1) juta rupiah setiap pemancarnya. komunitas petani.000 situs yang berisi berbagai rangkaian & informasi untuk membuat pemancar FM sendiri. 3.google. 15. tentunya dengan kebutuhan yang ada jarak pancarnya maupun pembebanan frekuensi dapat menjadi sangat terbatasi. Tapi memang tidak sebaik FM untuk menyiarkan lagu-lagu & entertainment. salah satunya adalah buatan Andik yang dapat dilihat rangkaian & disain papan rangkaiannya di RBS Skematik .5-9. Rangkaiannya tersedia dengan mudah di Internet. di Jogya.4MHz. ada baiknya mencari di Internet. atau radio rakyat ini. anda akan menemukan 36.75-5.1-15.95-6.45-21. juga terdapat alokasi band radio siaran pada frekuensi 2. 3. Band FM menjadi favorit karena kualitas pancarannya yang baik. tapi banyak kosong di kota-kota kecil maupun di kota kabupaten / kecamatan. dan tertinggi pada 25. pemancar-pemancar FM buatan sendiri ini dapat dibuat dengan mudah & murah.3MHz. komunitas pengrajin yang lingkupnya terbatas dan terkadang tidak terakomodasi oleh radio siaran swasta niaga.com. Band AM & SW praktis tidak digunakan.300-2. di banyak kota-kota kecil di Indonesia. jika sebuah frekuensi di duduki oleh sebuah transmitter yang sangat kuat (misalnya 10-30 KiloWatt) dengan jangkauan puluhan mungkin ratusan km. sebetulnya sangat sederhana. Tapi bayangkan jika atas kesepakatan bersama. Pada saat ini band FM menggunakan standar spasi 350KHz antar pemancarnya untuk mencapai kualitas suara yang sangat baik. seperti tertuang pada Bab VI Hak Atas Kebebaskan Informasi. Konsep ini di sebut frekuensi re-use. Pasal 20 dari Ketetapan MPR XVII/MPR/1998. kekuatan maksimum pemancar yang digunakan di batasi misalnya 10 Watt. Dengan semakin banyaknya radio pemancar komunitas. maka 8889MHz dapat dengan mudah di penuhi oleh 10 pemancar sekaligus.000 pemancar radio komunitas seluruh Indonesia. Tentunya masing-masing pemancar harus mentaati batasan daya yang di sepakati.000 pemancar radio komunitas. Tidak heran. frekuensi tersebut dapat dipakai ulang oleh pemancar yang berlainan. 24 July 2007 ) Basics of Stereo Multiplexing Written by BuSan Friday. Jika di Indonesia ada sekitar 4000-an kecamatan (400-an kabupaten). jika kita mengorbankan 88-90MHz untuk pemancar radio komunitas. Tanpa saling mengganggu antar pemancar walaupun ke duanya memancar pada saat yang bersamaan. Konsep frekuensi re-use ini merupakan jawaban yang telak mementahkan argumentasi bahwa terjadi pemborosan frekuensi oleh radio pemancar komunitas. dan daya tersebut harus rendah supaya frekuensi re-use dapat dilakukan sebanyak mungkin. dengan ketinggian antenna sekitar 37 meter. Artinya apa? Dalam setiap jarak 10-20 km. dengan menurunkan index modulasi sehingga sebuah pemancar hanya menggunakan lebar band 200KHz. akan lebih banyak lagi bangsa Indonesia terpenuhi hak asasinya untuk dapat berkomunikasi dan memperoleh informasi untuk mengembangkan pribadi dan lingkungan sosialnya. sebuah kekuatan warga yang sangat besar yang tidak dapat di kesampingkan begitu saja oleh siapapun. Dengan kualitas suara yang baik (lebar band 350KHz setiap pemancar). akan di peroleh jarak pancar sekitar 5-10 km untuk dapat diterima di radio biasa yang tidak terlalu baik. Last Updated ( Tuesday. maka kita akan memperoleh alokasi untuk 5 pemancar sekaligus yang dapat bekerja bersamaan dalam sebuah kecamatan atau kelurahan dalam radius pancaran 10-20 km. Jika kita sepakat untuk menurunkan sedikit kualitas suara. aparat & wakil rakyat Indonesia masih ingat kesepakatan yang tertuang di atas. Konsekuensinya kita akan melihat 40. Dengan perhitungan propagasi adanya redaman di udara di 89MHz.pemancar tersebut. Semoga para eksekutif. 17 November 2006 . radio komunitas di amerika serikat sana di kenal sebagai pemancar FM berdaya rendah (low power). dapat dengan mudah kita membangun 20. Masalah lain yang perlu diperhitungkan juga adalah lebar band setiap pemancar. konsekuensinya alokasi frekuensi untuk pemancar di FM 88-108MHz menjadi sekitar 56-57 pemancar saja. pada frekuensi yang bersamaan. an op to transmit the L signal signal hidden somewhere higher than 0-15 KHz as part of the Stereo MPX signal. The higher frequency signal use is then called a Subcarrier. AM stations on the other hand are s KHz away from each other. Lightning etc are normally AM sources.e Right channel. If the receiver adds the L+R signal to the L-R signal. To regenera Channel.e. . Here we need to know two rules of Electronics : A) If you Amplitude Modulate a carrier of frequency fc Hz with Audio that has a bandwidth of 0-fmaxa Hz. it gets 2R. the Right signal amplified two times. In this FM Receiver would not notice any change. One would obviously not sound stage but on the other hand.e. B) The carrier frequency has to be at least 2 times the highest modulation frequency otherwise one does not h "samples" to reproduce the original signal. Summing and signals is easy with just simple Summing and Difference Amps. As the 0-15 KHz part of the MPX signal has already been occupied by the L+R signal. it would just play back the L+R signal. FM scientists found a more elegant approach that above. i. one now has to think u accommodating the L-R signal somewhere higher up in the bandwidth allocated for the MPX signal. L+R. it gets 2L i. Add L+R to -L and extract a R signal i.e. As older FM Radios only looked at the 0-15KHz part of the incoming FM signal. All the Audio information we had from 0 to fmaxa in the side bands from fc-fmaxa to fc+fmaxa. However. one would not loose any information from the L and the R channels.One of the promises of FM in the early days was the possibility of transmission in Stereo. rejected by FM receivers • More distance on the dial between FM stations FM Stations are 200 KHz away from each other. They decided to transmit an L-R signal as MPX signal. FM Bandwidth = 2 ( deviation + input signal frequency). The process of combining multiple signals on composite signal in such a way that the the original signals can be reconstituted by the receiver is called Mult One of the design goals was that the Stereo Multiplexed signal ( MPX) was to be compatible with the earlier Radios. this gives lots of room to put in the Audio spectrum twice ov more information in one channel without infringing on the other channel. As designed to handle Audio from 20Hz to 15 KHz. it was a straightforw that the lower part of the Stereo MPX should contain the sum of the Left and Right channels i. the Left signal amplified two receiver subtracts L+R and the L-R signal. This stemmed direc of FM's properties : • Better rejection than AM of spurious signals from your girlfriend's Hair Dryer Electrical sparks. So the FM Gurus set out to decide on standards on how to use the available bandwidth to broadcast both the L Right Channel of a Stereo Broadcast on the same FM Channel. y side bands extending from fc-fmaxa right up to fc+fmaxa. The receiver could then direct the L Channel from the MPX signal directly to the Left Amplifier. One way R signal is to actually use it to amplitude modulate a higher frequency signal. the Receiver would convert L into -L by an Inverting Amp. If one is already transmitting a L+R channel and one finally needs the L and the R channels separately. the Rece this 19KHz signal to generate a 38KHz signal using a simple frequency doubler. The 38 KHz signal generated by the Receiver will be in phase with pilot tone. the Mono Receiver will not decode it. since it would be too hard for all FM Receivers to exactly generate the same frequency and phase. we would need an actual carrier to carry the signal across the airwaves application. the Subcarrier frequency for the L-R channel has to be higher th otherwise. It contains no information. As 38KHz is more than 2x 1 good enough to act as our AM Subcarrier. The MPX Coder / Transmitter wil A) Add Left and Right signals to get a L+R signal. it can act as the signal informing the Receiver that the signal is Stereo. B) Generate a Pilot Tone of 19 KHz. it can switch in its Stereo decoding circuitry. If the receiver r tone. By now we have finalized out how to generate the Stereo Multiplex signal.1) Use an AM Subcarrier of 38 KHz in phase with the 19 KHz Pilot.e.3) Cancel out the 38 KHz Subcarrier D) Add up A.2) Amplitude Modulate the 38 KHz AM Subcarrier with a L-R signal C. so it should decode information higher than 15 KHz as that's not just Audio. More-over. B and C above to get the complete MPX Signal. the receiver needs to regenerate the AM Subcarrier at the exact frequency and phase as the original extract the side band information. . you cannot reproduce signals of 15KHz. the Receiver wou exactly regenerate the missing AM SubCarrier. 23 KHz to 53 KHz. One simple way is to also transmit a sine wave as a pilot tone at a fixed frequency. We cannot just use a local oscillator at the agreed upon AM Subcarrier freq Receivers. Also from (B) abo Subcarrier frequency is below 30KHz. from (A) above. so in case the Transmitter uses a 38 KHz Subcarrier in phase with the Pilot Tone. we need the AM Subcarrier only to modulate it and generate the side bands storing L-R informat carrier is the FM frequency we are transmitting on. else it just treats the incoming signal as Mono. C. C. Use the above MPX signal to Frequency Mod in the 88-108 MHz band. ( Such an AM signal is called a Double Side Band Suppr DSBSC Signal ) However. If we choose a Pilot To in the 18-22 KHz range. so they would not overlap w Tone or the L+R information. it is the L-R signal h somehow. There is also the issue of somehow adding a tiny signal in the MPX signal that informs the Receiver that the i is in Stereo. Now comes a stroke of genius : Suppose we c Tone of 19 KHz. For an AM transmission. the side bands of the L-R channel will overlap the L+R signal.If we intend to transmit Audio upto 15 KHz. The Pilot tone has to be higher than 15 KHz as that part is used up by the L+R signal. It actually energy in it and we would just be wasting our FM transmitter's power by sending out an internal signal that co information. The Side Bands would be from 38-15 KHz to 38+15 KHz i. We actually do not need the AM Subcarrier at fc to be transmitted at all. How to actually generate the MPX si transmitting end? The hard way is to actually follow faithfully the above instructions in building up the MPX Summing Amp to add the L + R. Build a Double Balanced Mo feed it the L-R signal. from WEFM Chicago and WGFM Schenectady. 24. an interesting page with many more facts. Above information from FM Broadcasting Chronology. Armstrong demonstrates frequency modulation to executives and engineers of RCA July 18. 35. It also generates a lot of harmonics but they can be filtered away.2) Decoder will generate a 38 KHz signal by frequency doubling the received 19 KHz Pilot.8 MHz. 1940 : The first advertising contract for FM broadcasts was signed by the Longines Watch Company Jan. Switching channels as above makes a L+R signal and a D channel centered around 38 KHz. on this date the FCC received its first notificati regular operation.3) Decoder will demodulate the signals from 23KHz to 53 KHz to get back a L-R channel B. the ease of t method lead to it being accepted as the standard ? Rapidly sampling L and R signals alternatively is called Ti Multiplexing (TDM). I can just ask SPICE to scope out the expression "V(Left . However I soon realized that SPICE can show virtual waveforms based on expressions.) that rapidly switching between L and R channels at 38 KHz do hard work and is a near equivalent to the hard way. In case I n L+R signal.A Mono receiver will : A) React only to signals below 15 KHz so it would decode L+R A Stereo Receiver will : A) Sense the 19 KHz pilot signal so it will switch in the Stereo Decoder Circuitry B. 8. It will ensure t signal is in phase with the 19 KHz signal. the above scheme looks workable. 1945 : FCC announces allocations proposals.000 watts receivers in the world at that time !!! Dec. Winter 1933 : Edwin H. All that is the Pilot Tone. 1939 : First day of regular programming for W2XMN Alpine (Armstrong). 15. Have a Difference Amp create the L-R signal. I do not really need to simulate a mixer.1) Decoder will extract the L+R signal from the MPX signal B. Make a 38 KHz oscillator to feed the Double Balanced Modulator. with 84-88 MHz reserved noncommercial FM broadcasting. Build a final summing amp that will add L+R. Now comes the interesting part. I have just begun playing aro Micro-Cap SPICE so at first I thought I would simulate a complete Stereo Multiplexor and show the various w generated. Apr. 1961 : FM stereo broadcasting is authorized to begin. Maybe the FM MPX standards were actually decided upon the other way around. 42. the 19 KHz pilot and the modulated final MPX signal. It can however be shown ( not by me though . 1961 : Broadcasting reports the FCC approves stereo multiplex standards June 1.4) Decoder will use the L+R and the L-R signals to regenerate the original L and R signals OK. Now we really need some graphics to illustrate the various waveforms involved. B. moving FM to 84-108 MHz. Digitally divide the to get a 19 KHz pilot tone. . We wanted a smooth sine wave kind of thing whose amplitud between the instantaneous amplitude of the Left and the Right channels.V In the first graph above.. The chopped up basic MPX signal is shown in wanted is not really a chopped signal as above. all I need to do is ask for a plot of "IF(V(Chopper). we follow the Right channel. The chopped up waveform as above edges so it will have a lot of odd harmonics. When the chopper is high Left channel and when it is low. Let's run a harmonic analysis on the chopped MPX signal.4 Vpp.. to show the chopped up MPX signal. V(Left).. with 2 Vpp. The R is a 8 KHz signal at 1.. We have a Left Audio signal of a single Sine wave tone at 2 KHz .Similarly. We have a digital chopper in Blue running at 38 KHz. Besides. KHz signal transforms to signals of 36 and 40 KHz. BA1404 have used an external LPF on the MPX signal for a much better performance. centered around 38 KH bands of the L-R signal. we need a Low Pass Filter around 57 KHz to remove the harmonics in the MPX signal. Extra energy in the wrong place means that we waste transmitter po also limit the maximum modulation deviation for the signals we really want to transmit. The original 8 KHz signal transforms to 38-8 = 30 KHz and 38+8 = 46 KHz signals. At the bottom is the harmonic analysis of the sam can see the 2 KHz and 8 KHz signals in the L+R part of the MPX.Here we show on the top. The 38 KHz carrier signal is most obvious in its absence As predicted. 266 KHz an extraneous harmonics dirty up the signal. They also use up spa have been used for other signals as part of the MPX bundle. the BA1404 chip works by using exactly the same chopping method described above. 190. centered around 114. we also see strong odd harmonics of the DSBSC signal. I am not sure how this makes the final A Clearly. ( Incidentally . the original chopped MPX signal. Further on we see. they lowered costs by eliminating the vital MPX LPF) . Unfortunately most kit manufa to not take the design trouble. it is high in 3rd and harmonics. At the bottom. If we to . we read the L signal once and the R signal once i. We can see much lower h around 114 KHz while all other higher harmonics have been almost totally suppressed. we see the harmonic content of the MPX signal after the LPF.e. we cannot use it for FM MPX because shifting the 38 KHz carrier upwards would use up more ban would have a larger wasted gap between the L+R and the L-R part of the signal. As expected. The first thought that comes to mind is to increase the sampling frequency.The smoother red line in the top chart is the MPX signal after it has gone through a very simple multistage RC Filter. The examples above show that we use the Left signal in case the Chopper signal is high and Right signal in c Chopper signal is low. While this idea works ocassions. there are two states odd looking wave that appears as if it is made up of square wave segments. However the LPF has enormous phase shifts. So what we need to do is intr between the 38 KHz samples. A better LC LPF could have been used instead for be One way of looking at the MPX generation method is that it is simply sampling the L and the R channels at 3 standard method of reducing extraneously generated harmonic content in sampled Analog signals is to increa of samples. more about which follows later. In each period. we see that what w was a smoother sine-like curve with amplitude oscillating between the instantaneous L and R signals. If we look at the red curve showing the same chopped MPX signal passed through a LPF. so we multiply the at 45 degrees by 0. V(Left). V(Left). + (IF((tm2>=315 AND tm2<=360).V(Left)*.V(Right). + (IF((tm2>=225 AND tm2<315).V(Right)) we now use following snippet from SPICE : .V(Right)*.293.293. + (IF((tm2>=135 AND tm2<180). This would be much lower in harm square wave because it looks much closer to the signal that we really want. SIN(45) is 0. . If we break one period into 8 time intervals.293. V(Left)*. each bit will be 45 degrees.samples. So instead of old algorithm of just switching between full L and full R.V(Right)*.707+ V(Left)*.707.707. we would get a "Digital Sine" wave. IF(V(Chopper).707+ V(Right)*. + (IF((tm2>=45 AND tm2<135).707+ V(Right)*. let's say 8 times per period.define WTD_MPX IF((tm2>=0 AND tm2<45). + (IF((tm2>=180 AND tm2<225).707+ V(Left)*.293. + 0))))))))))) We need only 4 different states to generate the new waveform. I was amazed at the reduction of Harmonic content caused by this simple weighted oversampling technique !! . it contains a lot of odd harmonics. Even better results would have been possible if we oversampled at a higher rate i. the Right Audio..e. we would get a final Pilot tone wit reduced harmonics. with the raw oversampled signal looking better than the basic MPX throug find that the first serious harmonic in the weighted MPX signal is around 266 KHz ie the 7th harmonic. leading to better channel separation. So far. The Bottom part shows the harmonic analysis of the MPX and the weighted MPX signals. . Professional Stereo Coders do use 8 State s Exactly the same weighted sampling technique can be used to generate the 19 KHz Pilot..The top part shows the Left Audio. Let's try and simulate a bit of musi a basic chopped MPX with a low pass filter. Basic chopped MPX and oversampled weighted MPX si of each other. we have used only Sine Wave inputs to study the MPX waveform. but with greater stability ( Locked to a Cr than an analog generated Sine wave. a gentler MPX filter with lower phase shifts in the 0-53KHz can no designed. used 8 states and weight s KHz instead of the 4 states weight sampled at 304 KHz as shown. If we use a digitally Square wave like in the BA1404 and many kits.. As th away from the 0-53 KHz signals we want.. The oversampling are dramatic indeed. If we tried to create a "Dig Wave" by making weighted samples at 45 degree intervals in each period. much closer to the sine wave we really wanted. If you play a CD with 20 KHz information. a stereo receiver would be within its rights to reproduce a 8 KHz tone during decoding p have mysteriously created a 8 KHz signal from a clipped 10 KHz signal !! This is called "Aliasing". It will create harmonic distortion at 2x.The Black line in the top half is the basic chopped MPX signal passed through a LPF. We can see that the area between 15 KHz and 23 KHz is not blank as we would have imagined. Suppose a filter in the Audio phase around 30 KHz. One of the interesting effects of the MPX process is non-harmonic distortion in case of clipping. but no Pilot Tone has b the Harmonic Analysis graph. When we a . It is also interesting to study the effects of phase shifts within the Audio Chain. Such sharp filters are called brickwall filters. it will shift part of the L-R signal. maybe because the sharply stop frequencies past their cutoff frequency analogous to one driving a car into a brickwall. Let's call the phase shifted signal Lp-Rp. Obviously one needs pretty sharp 15 KHz Low Pass Filters on both Aud before they are fed into an MPX generator. Even a Bass drum (which we would assume contains only low frequencies) is actually rich harmonics at the moment when the drum is struck. 3x etc harmonic will fall at 30 KHz. Lets analyze comes about. it will ruin the due to overlapping the 19 KHz Pilot. All musical i create harmonics. Suppose we have a 10 KHz signal that is clipped. followed by th between 23 to 53 KHz. we can easily make out the L+R region between 0 and 15 KHz. But that's right in the territory of the L-R signal !! As a 30 KHz signal is 8 KHz 38 KHz AM carrier. 81 -0.79 .0115/2 of crosstalk ie -44. Similarly.05 -50. Channel Separation.1 dB down.988 L + 0.0115 R instead of 2L We see that the MPX process has added 0. how that comes about.988 L . So if t electronics gave you 50 dB of separation. a change in gain over the 0-53 KHz region will also negatively affect the Channel Separation.0. This means that the signal is less by a factor of 0.01 -64. This will make you loose about 6-10 dB of separation.0114 R Hey. The Decoder will add the L+R and the L-R 2L But we will get (1 L + 1 R) + (0. To try and filter that by about 20 dB will need a filter that would probably cause a around 50-70 degrees around 45 KHz upwards. As the phase shift could also have happened in the other stages following the Stereo Multiplexer (Transmitter or even at the Receiver). Gain Change dB Channel Separation dB -0. we see that R and Rp will not cancel each other exactly even if they are of Amplitude. suppose we turn up the volume by 2/1. one would tend to unnecessarily place the whole blame on the Multiplexer.79 -0.1 dB from there to 53 So the L-R part of the MPX signal will be 0. the nearest unwanted harmonic in the signal harmonic around 114 KHz.Lp-Rp to try and get the 2L signal. (Loss of Channel Separation) So to compensate for the less L. we wanted a theoretical signal of 2L over here. A small component of the Right channel will remain in the Left Channel.9885531 times.988 R) = 1.988 Suppose we inject enough Audio to get 1L + 1R in the L+R part. Loss of S/N) B) Puts in a bit of R into the L channel. We get 2L + 0.1 -44. calculated the same way as above.79 dB Here's a small table re gain change vs. But we see that loss of gain in the 19-53 KHz part of the A) Reduces our L ( Receiver needs more Audio Gain. Similarly a small bit of channel will remain in the right channel. So a phase shift in the Audio MPX Chain causes loss of Channel This is a serious issue : If you dont use oversampling techniques. Suppose we have a Multiplexing stage that has 0 dB gain from 0 to 19KHz and then -0. the MPX filter will reduce that to 40-44 dB. 2 -0. Phew As changes in Gain and Phase shift are frequency dependant. As an MPX signal is not really signal.05 db across 0 . Hence a Transmitter signal will travel less distance in Stereo than wh transmitting in Mono. degrading sharply at both the low high end.34 -12.53 KHz for separation better than 50 dB.80 -18. the Harmonic analyses would show a blip at the 19 KHz point. The MPX signals would look quite the same a shown above.-0. The decoder circuits in the receive contribute extra noise.81 -24. and Voila. All you can do now is feed the MPX signal to the transmitter instead of the Mono A had before.91 So looks like we need a gain change below 0.5 -1 -2 -3 -4 -38. Put another way. the Channel Separation of a Stereo Decoder is f dependent. you cannot Compress/Limit/Equalise/Pre-emphasise an MPX signal. we now have more noise than receiving the same signal in Mono. A Mono Transmitter would be set up to allow maximum allowed deviation on the L+R signal it is transmitting. as MPX signals have more bandwidth than a Mono signal. The Stereo Transmitter would have to be turned down a bit as it also needs t deviation caused by the L-R signals. All above tend to increase the noise. 17 November 2006 . (Visit USENET below for more info) The last step is to add in about 10% of the 19 KHz Pilot Tone. You are transmitting in glorious STerEO !!! FM Transmitters Written by BuSan Friday. A Stereo transmitter has to transmit various parts of the MPX signal and has to be set such that it does not ove the sum of all the MPX signal. All the audio processing should h before the MPX process.81 -15. This is mainly caused by the noise in the As the noise in the 23 KHz to 53 KHz segment is also brought down to the audible 0-15 KHz region by the d process. if you need to start transmitting in Stereo. On top of that. the station h modulation with MPX to remain in the deviation limits. you need a more powerful A Stereo transmission tends to sound noisier that a Mono signal. Normally Channel Separation is best in the 200-8000 Hz range.77 -30. Indirect FM involves directly altering the phase of the carrier based on the input (this is actually a form of direct phase modulation. which is depicted in figure 2. the varactor diode modulator can only be used in limited applications. The use of a crystal oscillator means that the output waveform is very stable. The first method can be accomplished by the use of a reverse biased diode. since the capacitance of such a diode varies with applied voltage. R1 and R2 develop a DC voltage across the diode which reverse biases it. decrease the diode capacitance and thus increase the oscillation frequency. Thus. Figure 1 shows a direct frequency modulator which uses a varactor diode. Direct modulation is usually accomplished by varying a capacitance in an LC oscillator or by changing the charging current applied to a capacitor.MODULATORS There are two types of FM modulators .direct and indirect. Similarly. negative inputs decrease the oscillation frequency. . The second method of direct FM involves the use of a voltage controlled oscillator. The voltage across the diode determines the frequency of the oscillations. A varactor diode is specifically designed for this purpose. Direct FM involves varying the frequency of the carrier directly by the modulating input. but this is only the case if the frequency deviations are kept very small. Positive inputs increase the reverse bias. This circuit deviates the frequency of the crystal oscillator using the diode.